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Diffstat (limited to 'net/asterisk-bristuff/files/patch-channels::chan_oss.c')
-rw-r--r--net/asterisk-bristuff/files/patch-channels::chan_oss.c1167
1 files changed, 0 insertions, 1167 deletions
diff --git a/net/asterisk-bristuff/files/patch-channels::chan_oss.c b/net/asterisk-bristuff/files/patch-channels::chan_oss.c
deleted file mode 100644
index ef8cfc11d711..000000000000
--- a/net/asterisk-bristuff/files/patch-channels::chan_oss.c
+++ /dev/null
@@ -1,1167 +0,0 @@
-
-$FreeBSD$
-
---- channels/chan_oss.c
-+++ channels/chan_oss.c
-@@ -13,6 +13,8 @@
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
-+ *
-+ * FreeBSD changes by Luigi Rizzo, 2005.04.18
- */
-
- #include <asterisk/lock.h>
-@@ -54,21 +56,30 @@
- #endif
-
- /* Lets use 160 sample frames, just like GSM. */
--#define FRAME_SIZE 160
-+/* this corresponds to 20ms of audio. */
-+#define FRAME_SIZE 160 // was 160
-
--/* When you set the frame size, you have to come up with
-- the right buffer format as well. */
-+/*
-+ * When you set the frame size, you have to come up with
-+ * the right buffer format as well.
-+ * OSS lets you define a 'block' size (which should be a power of 2,
-+ * which power is specified in the lower 16 bits) and the number of
-+ * blocks allowed in the buffer (to avoid that the queue grows too large).
-+ * The latter is specified in the top 16 bits.
-+ * We use a block of 64 bytes (0x6), 5 blocks make a frame each sample
-+ * being 2 bytes, and we make room to store two buffers.
-+ * XXX the '10' is magic
-+ */
-+
-+#define N_BLOCKS (buffersize * 5 * 2)
- /* 5 64-byte frames = one frame */
--#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
-+#define BUFFER_FMT (N_BLOCKS << 16) | (0x0006);
-
- /* Don't switch between read/write modes faster than every 300 ms */
--#define MIN_SWITCH_TIME 600
-+#define MIN_SWITCH_TIME 300
-
--static struct timeval lasttime;
-
- static int usecnt;
--static int silencesuppression = 0;
--static int silencethreshold = 1000;
-
-
- AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-@@ -78,16 +89,15 @@
- static char *tdesc = "OSS Console Channel Driver";
- static char *config = "oss.conf";
-
--static char context[AST_MAX_EXTENSION] = "default";
-+static char default_context[AST_MAX_EXTENSION] = "default";
- static char language[MAX_LANGUAGE] = "";
--static char exten[AST_MAX_EXTENSION] = "s";
-+static char oss_exten[AST_MAX_EXTENSION] = "s";
-
--static int hookstate=0;
-
--static short silence[FRAME_SIZE] = {0, };
-
- struct sound {
- int ind;
-+ char *desc;
- short *data;
- int datalen;
- int samplen;
-@@ -96,136 +106,178 @@
- };
-
- static struct sound sounds[] = {
-- { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
-- { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
-- { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
-- { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
-- { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
-+ { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
-+ { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
-+ { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
-+ { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
-+ { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
-+ { -1, NULL, 0, 0, 0, 0 }, /* end marker */
- };
-
--/* Sound command pipe */
--static int sndcmd[2];
-+
-
- static struct chan_oss_pvt {
- /* We only have one OSS structure -- near sighted perhaps, but it
- keeps this driver as simple as possible -- as it should be. */
-+ /*
-+ * cursound indicates which in struct sound we play. -1 means nothing,
-+ * any other value is a valid sound, in which case sampsent indicates
-+ * the next sample to send in [0..samplen + silencelen]
-+ * nosound is set to disable the audio data from the channel
-+ * (so we can play the tones etc.).
-+ */
-+ int sndcmd[2]; /* Sound command pipe */
-+ int cursound; /* index of sound to send */
-+ int sampsent; /* # of sound samples sent */
-+ int nosound;
-+
-+ int total_blocks; /* total blocks in the output device */
-+ int sounddev;
-+ enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
-+ int autoanswer;
-+ int autohangup;
-+ int hookstate;
-+ struct timeval lasttime; /* last setformat */
-+
-+ int silencesuppression;
-+ int silencethreshold;
-+ char device[64]; /* device to open */
-+
-+ pthread_t sthread;
-+
- struct ast_channel *owner;
- char exten[AST_MAX_EXTENSION];
- char context[AST_MAX_EXTENSION];
--} oss;
-+} oss = {
-+ .cursound = -1,
-+ .sounddev = -1,
-+ .duplex = M_UNSET, /* XXX check this */
-+ .autoanswer = 1,
-+ .autohangup = 1,
-+ .silencethreshold = 1000,
-+};
-
--static int time_has_passed(void)
-+/*
-+ * returns true if too early to switch
-+ */
-+static int too_early(struct chan_oss_pvt *o)
- {
- struct timeval tv;
- int ms;
- gettimeofday(&tv, NULL);
-- ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
-- (tv.tv_usec - lasttime.tv_usec) / 1000;
-- if (ms > MIN_SWITCH_TIME)
-+ ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 +
-+ (tv.tv_usec - o->lasttime.tv_usec) / 1000;
-+ if (ms < MIN_SWITCH_TIME)
- return -1;
- return 0;
- }
-
--/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
-- with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
-- usually plenty. */
--
--static pthread_t sthread;
--
--#define MAX_BUFFER_SIZE 100
--static int buffersize = 3;
--
--static int full_duplex = 0;
--
--/* Are we reading or writing (simulated full duplex) */
--static int readmode = 1;
--
--/* File descriptor for sound device */
--static int sounddev = -1;
--
--static int autoanswer = 1;
--
--#if 0
--static int calc_loudness(short *frame)
-+/*
-+ * Returns the number of blocks used in the audio output channel
-+ */
-+static int
-+used_blocks(struct chan_oss_pvt *o)
- {
-- int sum = 0;
-- int x;
-- for (x=0;x<FRAME_SIZE;x++) {
-- if (frame[x] < 0)
-- sum -= frame[x];
-- else
-- sum += frame[x];
-+ struct audio_buf_info info;
-+ static int warned=0;
-+ if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
-+ if (!warned) {
-+ ast_log(LOG_WARNING, "Error reading output space\n");
-+ warned++;
- }
-- sum = sum/FRAME_SIZE;
-- return sum;
-+ return 1;
-+ }
-+ if (o->total_blocks == 0) {
-+ ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n",
-+ info.fragstotal,
-+ info.fragsize,
-+ info.fragments);
-+ o->total_blocks = info.fragments;
-+ }
-+ return o->total_blocks - info.fragments;
- }
--#endif
-
--static int cursound = -1;
--static int sampsent = 0;
--static int silencelen=0;
--static int offset=0;
--static int nosound=0;
-+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
-+{
-+ /* Write an exactly FRAME_SIZE sized of frame */
-+ int res;
-+ static int errors = 0;
-
--static int send_sound(void)
-+ /*
-+ * nothing spectacular.
-+ * If the buffer is full just drop the extra, otherwise write
-+ */
-+ res = used_blocks(o);
-+ if (res > 10) { /* no room to write a block */
-+ errors ++;
-+ if (errors == 0)
-+ ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, errors);
-+ return 0;
-+ }
-+ errors = 0;
-+ res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
-+ return res;
-+}
-+
-+/*
-+ * handler for 'sound writable' events from the sound thread.
-+ * Builds a frame from the high level description of the sounds,
-+ * (tone+silence) and passes it to the audio device.
-+ */
-+static int send_sound(struct chan_oss_pvt *o)
- {
- short myframe[FRAME_SIZE];
-- int total = FRAME_SIZE;
-- short *frame = NULL;
-- int amt=0;
-- int res;
-- int myoff;
-- audio_buf_info abi;
-- if (cursound > -1) {
-- res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
-- if (res) {
-- ast_log(LOG_WARNING, "Unable to read output space\n");
-- return -1;
-- }
-- /* Calculate how many samples we can send, max */
-- if (total > (abi.fragments * abi.fragsize / 2))
-- total = abi.fragments * abi.fragsize / 2;
-- res = total;
-- if (sampsent < sounds[cursound].samplen) {
-- myoff=0;
-- while(total) {
-- amt = total;
-- if (amt > (sounds[cursound].datalen - offset))
-- amt = sounds[cursound].datalen - offset;
-- memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
-- total -= amt;
-- offset += amt;
-- sampsent += amt;
-- myoff += amt;
-- if (offset >= sounds[cursound].datalen)
-- offset = 0;
-- }
-- /* Set it up for silence */
-- if (sampsent >= sounds[cursound].samplen)
-- silencelen = sounds[cursound].silencelen;
-- frame = myframe;
-- } else {
-- if (silencelen > 0) {
-- frame = silence;
-- silencelen -= res;
-- } else {
-- if (sounds[cursound].repeat) {
-- /* Start over */
-- sampsent = 0;
-- offset = 0;
-- } else {
-- cursound = -1;
-- nosound = 0;
-- }
-- }
-+ int ofs = 0;
-+ int l_sampsent = o->sampsent;
-+ int l;
-+ struct sound *s;
-+
-+ if (o->cursound < 0) /* no sound to send */
-+ return 0;
-+ s = &sounds[o->cursound];
-+ /*
-+ * prepare a frame
-+ */
-+
-+ for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
-+ /* take chunks of sound and data until the buffer is full */
-+ l = s->samplen - l_sampsent; /* sound available */
-+ if (l > 0) {
-+ if (l > FRAME_SIZE - ofs)
-+ l = FRAME_SIZE - ofs;
-+ if (0)
-+ ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n",
-+ l_sampsent, l, s->samplen, ofs);
-+ bcopy(s->data + l_sampsent, myframe + ofs, l*2);
-+ l_sampsent += l;
-+ } else { /* no sound, maybe some silence */
-+ static short silence[FRAME_SIZE] = {0, };
-+
-+ l += s->silencelen;
-+ if (l > 0) {
-+ if (l > FRAME_SIZE - ofs)
-+ l = FRAME_SIZE - ofs;
-+ if (0)
-+ ast_log(LOG_WARNING, "send_sound silence %d/%d of %d into %d\n",
-+ l_sampsent - s->samplen, l, s->silencelen, ofs);
-+ bcopy(silence, myframe + ofs, l*2);
-+ l_sampsent += l;
-+ } else { /* silence is over, restart sound if loop */
-+ if (s->repeat == 0) { /* last block */
-+ ast_log(LOG_WARNING, "send_sound last block\n");
-+ o->cursound = -1;
-+ o->nosound = 0; /* allow audio data */
-+ if (ofs < FRAME_SIZE) /* pad with silence */
-+ bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2);
-+ }
-+ l_sampsent = 0;
- }
-- if (frame)
-- res = write(sounddev, frame, res * 2);
-- if (res > 0)
-- return 0;
-- return res;
-+ }
- }
-- return 0;
-+ l = soundcard_writeframe(o, myframe);
-+ if (l > 0)
-+ o->sampsent = l_sampsent; /* update status */
-+ return 0; /* fake success */
- }
-
- static void *sound_thread(void *unused)
-@@ -235,41 +287,53 @@
- int max;
- int res;
- char ign[4096];
-- if (read(sounddev, ign, sizeof(sounddev)) < 0)
-+ if (read(oss.sounddev, ign, sizeof(ign)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
- for(;;) {
- FD_ZERO(&rfds);
- FD_ZERO(&wfds);
-- max = sndcmd[0];
-- FD_SET(sndcmd[0], &rfds);
-+ max = oss.sndcmd[0];
-+ FD_SET(oss.sndcmd[0], &rfds);
- if (!oss.owner) {
-- FD_SET(sounddev, &rfds);
-- if (sounddev > max)
-- max = sounddev;
-+ FD_SET(oss.sounddev, &rfds);
-+ if (oss.sounddev > max)
-+ max = oss.sounddev;
- }
-- if (cursound > -1) {
-- FD_SET(sounddev, &wfds);
-- if (sounddev > max)
-- max = sounddev;
-+ if (oss.cursound > -1) {
-+ FD_SET(oss.sounddev, &wfds);
-+ if (oss.sounddev > max)
-+ max = oss.sounddev;
- }
- res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
- if (res < 1) {
- ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
- continue;
- }
-- if (FD_ISSET(sndcmd[0], &rfds)) {
-- read(sndcmd[0], &cursound, sizeof(cursound));
-- silencelen = 0;
-- offset = 0;
-- sampsent = 0;
-+ if (FD_ISSET(oss.sndcmd[0], &rfds)) { /* read which sound to play from the pipe */
-+ int i, what;
-+
-+ read(oss.sndcmd[0], &what, sizeof(what));
-+ for (i = 0; sounds[i].ind != -1; i++)
-+ if (sounds[i].ind == what) {
-+ oss.cursound = i;
-+ oss.sampsent = 0;
-+ oss.nosound = 1; /* block other audio */
-+ ast_log(LOG_WARNING, "play %s\n", sounds[i].desc);
-+ break;
-+ }
-+ if (sounds[i].ind == -1)
-+ oss.cursound = -1;
-+ ast_log(LOG_WARNING, "cursound %d samplen %d silencelen %d\n",
-+ oss.cursound, oss.cursound >=0 ? sounds[oss.cursound].samplen : -1,
-+ oss.cursound >=0 ? sounds[oss.cursound].silencelen : -1);
- }
-- if (FD_ISSET(sounddev, &rfds)) {
-+ if (FD_ISSET(oss.sounddev, &rfds)) {
- /* Ignore read */
-- if (read(sounddev, ign, sizeof(ign)) < 0)
-+ if (read(oss.sounddev, ign, sizeof(ign)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
- }
-- if (FD_ISSET(sounddev, &wfds))
-- if (send_sound())
-+ if (FD_ISSET(oss.sounddev, &wfds))
-+ if (send_sound(&oss) < 0)
- ast_log(LOG_WARNING, "Failed to write sound\n");
- }
- /* Never reached */
-@@ -277,6 +341,20 @@
- }
-
- #if 0
-+static int calc_loudness(short *frame)
-+{
-+ int sum = 0;
-+ int x;
-+ for (x=0;x<FRAME_SIZE;x++) {
-+ if (frame[x] < 0)
-+ sum -= frame[x];
-+ else
-+ sum += frame[x];
-+ }
-+ sum = sum/FRAME_SIZE;
-+ return sum;
-+}
-+
- static int silence_suppress(short *buf)
- {
- #define SILBUF 3
-@@ -284,7 +362,7 @@
- static int silentframes = 0;
- static char silbuf[FRAME_SIZE * 2 * SILBUF];
- static int silbufcnt=0;
-- if (!silencesuppression)
-+ if (!oss.silencesuppression)
- return 0;
- loudness = calc_loudness((short *)(buf));
- if (option_debug)
-@@ -309,7 +387,7 @@
- /* Write any buffered silence we have, it may have something
- important */
- if (silbufcnt) {
-- write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
-+ write(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE);
- silbufcnt = 0;
- }
- }
-@@ -317,27 +395,55 @@
- }
- #endif
-
--static int setformat(void)
-+/*
-+ * reset and close the device if opened,
-+ * then open and initialize it in the desired mode,
-+ * trigger reads and writes so we can start using it.
-+ */
-+static int setformat(struct chan_oss_pvt *o, int mode)
- {
-- int fmt, desired, res, fd = sounddev;
-+ int fmt, desired, res, fd;
- static int warnedalready = 0;
- static int warnedalready2 = 0;
-+
-+ if (o->sounddev >= 0) {
-+ ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
-+ close(o->sounddev);
-+ o->duplex = M_UNSET;
-+ }
-+ fd = o->sounddev = open(o->device, mode |O_NONBLOCK);
-+ if (o->sounddev < 0) {
-+ ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n",
-+ strerror(errno));
-+ return -1;
-+ }
-+
-+ gettimeofday(&o->lasttime, NULL);
- fmt = AFMT_S16_LE;
- res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
- return -1;
- }
-- res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
--
-- /* Check to see if duplex set (FreeBSD Bug)*/
-- res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
--
-- if ((fmt & DSP_CAP_DUPLEX) && !res) {
-- if (option_verbose > 1)
-+ switch (mode) {
-+ case O_RDWR:
-+ res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
-+ /* Check to see if duplex set (FreeBSD Bug)*/
-+ res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
-+ if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
-+ if (option_verbose > 1)
- ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
-- full_duplex = -1;
-+ o->duplex = M_FULL;
-+ };
-+ break;
-+ case O_WRONLY:
-+ o->duplex = M_WRITE;
-+ break;
-+ case O_RDONLY:
-+ o->duplex = M_READ;
-+ break;
- }
-+
- fmt = 0;
- res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
- if (res < 0) {
-@@ -348,6 +454,7 @@
- desired = 8000;
- fmt = desired;
- res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
-+
- if (res < 0) {
- ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
- return -1;
-@@ -357,89 +464,54 @@
- ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
- }
- #if 1
-- fmt = BUFFER_FMT;
-+ /*
-+ * on freebsd, SETFRAGMENT does not work very well on some cards.
-+ * Better leave it out
-+ */
-+
-+ // fmt = BUFFER_FMT;
-+ fmt = 0x8; // 256-bytes fragment
- res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
- if (res < 0) {
- if (!warnedalready2++)
- ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
- }
- #endif
-+ /* XXX on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
-+ res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
-+ res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
-+ /* it may fail if we are in half duplex, never mind */
- return 0;
- }
-
-+/*
-+ * make sure output mode is available. Returns 0 if done,
-+ * 1 if too early to switch, -1 if error
-+ */
- static int soundcard_setoutput(int force)
- {
-- /* Make sure the soundcard is in output mode. */
-- int fd = sounddev;
-- if (full_duplex || (!readmode && !force))
-- return 0;
-- readmode = 0;
-- if (force || time_has_passed()) {
-- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-- /* Keep the same fd reserved by closing the sound device and copying stdin at the same
-- time. */
-- /* dup2(0, sound); */
-- close(sounddev);
-- fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-- return -1;
-- }
-- /* dup2 will close the original and make fd be sound */
-- if (dup2(fd, sounddev) < 0) {
-- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-- return -1;
-- }
-- if (setformat()) {
-- return -1;
-- }
-+ if (oss.duplex == M_FULL || (oss.duplex == M_WRITE && !force))
- return 0;
-- }
-- return 1;
-+ if (!force && too_early(&oss))
-+ return 1;
-+ if (setformat(&oss, O_WRONLY))
-+ return -1;
-+ return 0;
- }
-
-+/*
-+ * make sure input mode is available. Returns 0 if done
-+ * 1 if too early to switch, -1 if error
-+ */
- static int soundcard_setinput(int force)
- {
-- int fd = sounddev;
-- if (full_duplex || (readmode && !force))
-- return 0;
-- readmode = -1;
-- if (force || time_has_passed()) {
-- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-- close(sounddev);
-- /* dup2(0, sound); */
-- fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-- return -1;
-- }
-- /* dup2 will close the original and make fd be sound */
-- if (dup2(fd, sounddev) < 0) {
-- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-- return -1;
-- }
-- if (setformat()) {
-- return -1;
-- }
-+ if (oss.duplex == M_FULL || (oss.duplex == M_READ && !force))
- return 0;
-- }
-- return 1;
--}
--
--static int soundcard_init(void)
--{
-- /* Assume it's full duplex for starters */
-- int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
-- if (fd < 0) {
-- ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
-- return fd;
-- }
-- gettimeofday(&lasttime, NULL);
-- sounddev = fd;
-- setformat();
-- if (!full_duplex)
-- soundcard_setinput(1);
-- return sounddev;
-+ if (!force && too_early(&oss))
-+ return 1;
-+ if (setformat(&oss, O_RDONLY))
-+ return -1;
-+ return 0;
- }
-
- static int oss_digit(struct ast_channel *c, char digit)
-@@ -454,120 +526,81 @@
- return 0;
- }
-
-+/* request to play a sound on the speaker */
-+#define RING(x) { int what = x; write(oss.sndcmd[1], &what, sizeof(what)); }
-+
- static int oss_call(struct ast_channel *c, char *dest, int timeout)
- {
-- int res = 3;
- struct ast_frame f = { 0, };
- ast_verbose( " << Call placed to '%s' on console >> \n", dest);
-- if (autoanswer) {
-+ if (oss.autoanswer) {
- ast_verbose( " << Auto-answered >> \n" );
- f.frametype = AST_FRAME_CONTROL;
- f.subclass = AST_CONTROL_ANSWER;
- ast_queue_frame(c, &f);
- } else {
-- nosound = 1;
- ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
- f.frametype = AST_FRAME_CONTROL;
- f.subclass = AST_CONTROL_RINGING;
- ast_queue_frame(c, &f);
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(AST_CONTROL_RING);
- }
- return 0;
- }
-
- static void answer_sound(void)
- {
-- int res;
-- nosound = 1;
-- res = 4;
-- write(sndcmd[1], &res, sizeof(res));
--
-+ RING(AST_CONTROL_ANSWER);
- }
-
- static int oss_answer(struct ast_channel *c)
- {
- ast_verbose( " << Console call has been answered >> \n");
-- answer_sound();
-+ answer_sound(); /* XXX do we really need it ? considering we shut down immediately... */
- ast_setstate(c, AST_STATE_UP);
-- cursound = -1;
-- nosound=0;
-+ oss.cursound = -1;
-+ oss.nosound=0;
- return 0;
- }
-
- static int oss_hangup(struct ast_channel *c)
- {
-- int res = 0;
-- cursound = -1;
-+ oss.cursound = -1;
- c->pvt->pvt = NULL;
- oss.owner = NULL;
- ast_verbose( " << Hangup on console >> \n");
- ast_mutex_lock(&usecnt_lock);
- usecnt--;
- ast_mutex_unlock(&usecnt_lock);
-- if (hookstate) {
-- if (autoanswer) {
-+ if (oss.hookstate) {
-+ if (oss.autoanswer || oss.autohangup) {
- /* Assume auto-hangup too */
-- hookstate = 0;
-+ oss.hookstate = 0;
- } else {
- /* Make congestion noise */
-- res = 2;
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(AST_CONTROL_CONGESTION);
- }
- }
- return 0;
- }
-
--static int soundcard_writeframe(short *data)
--{
-- /* Write an exactly FRAME_SIZE sized of frame */
-- static int bufcnt = 0;
-- static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
-- struct audio_buf_info info;
-- int res;
-- int fd = sounddev;
-- static int warned=0;
-- if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
-- if (!warned)
-- ast_log(LOG_WARNING, "Error reading output space\n");
-- bufcnt = buffersize;
-- warned++;
-- }
-- if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
-- /* We've run out of stuff, buffer again */
-- bufcnt = 0;
-- }
-- if (bufcnt == buffersize) {
-- /* Write sample immediately */
-- res = write(fd, ((void *)data), FRAME_SIZE * 2);
-- } else {
-- /* Copy the data into our buffer */
-- res = FRAME_SIZE * 2;
-- memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
-- bufcnt++;
-- if (bufcnt == buffersize) {
-- res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
-- }
-- }
-- return res;
--}
--
--
-+/* used for data coming from the network */
- static int oss_write(struct ast_channel *chan, struct ast_frame *f)
- {
- int res;
-- static char sizbuf[8000];
-- static int sizpos = 0;
-- int len = sizpos;
-- int pos;
-+ int src;
-+
-+ // ast_log(LOG_WARNING, "oss_write size %d\n", f->datalen);
- /* Immediately return if no sound is enabled */
-- if (nosound)
-+ if (oss.nosound)
- return 0;
- /* Stop any currently playing sound */
-- cursound = -1;
-- if (!full_duplex) {
-+ oss.cursound = -1;
-+ if (oss.duplex != M_FULL) {
-+ /* XXX check this, looks weird! */
- /* If we're half duplex, we have to switch to read mode
- to honor immediate needs if necessary */
-- res = soundcard_setinput(1);
-+ res = soundcard_setinput(1); /* force set if not full_duplex */
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set device to input mode\n");
- return -1;
-@@ -583,21 +616,30 @@
- so just pretend we wrote it */
- return 0;
- }
-- /* We have to digest the frame in 160-byte portions */
-- if (f->datalen > sizeof(sizbuf) - sizpos) {
-- ast_log(LOG_WARNING, "Frame too large\n");
-- return -1;
-- }
-- memcpy(sizbuf + sizpos, f->data, f->datalen);
-- len += f->datalen;
-- pos = 0;
-- while(len - pos > FRAME_SIZE * 2) {
-- soundcard_writeframe((short *)(sizbuf + pos));
-- pos += FRAME_SIZE * 2;
-+ /*
-+ * we could receive a sample which is not a multiple of our FRAME_SIZE,
-+ * so we buffer it locally and write to the device in FRAME_SIZE
-+ * chunks, keeping the residue stored for future use.
-+ */
-+
-+ src = 0; /* read position into f->data */
-+ while ( src < f->datalen ) {
-+ static char buf[FRAME_SIZE*2];
-+ static int dst = 0;
-+ int l = sizeof(buf) - dst; /* how much room in the buffer */
-+
-+ if (f->datalen - src >= l) { /* enough to fill a frame */
-+ memcpy(buf + dst, f->data + src, l);
-+ soundcard_writeframe(&oss, (short *)buf);
-+ src += l;
-+ dst = 0;
-+ } else { /* copy residue */
-+ l = f->datalen - src;
-+ memcpy(buf + dst, f->data + src, l);
-+ src += l; /* but really, we are done */
-+ dst += l;
-+ }
- }
-- if (len - pos)
-- memmove(sizbuf, sizbuf + pos, len - pos);
-- sizpos = len - pos;
- return 0;
- }
-
-@@ -628,18 +670,15 @@
- ast_log(LOG_WARNING, "Unable to set input mode\n");
- return NULL;
- }
-- if (res > 0) {
-+ if (res > 0) { /* too early to switch ? */
- /* Theoretically shouldn't happen, but anyway, return a NULL frame */
- return &f;
- }
-- res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-- if (res < 0) {
-- ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
--#if 0
-- CRASH;
--#endif
-- return NULL;
-- }
-+
-+ res = read(oss.sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-+ // ast_log(LOG_WARNING, "oss_read() fd %d got %d\n", oss.sounddev, res);
-+ if (res < 0) /* audio data not ready, return a NULL frame */
-+ return &f;
- readpos += res;
-
- if (readpos >= FRAME_SIZE * 2) {
-@@ -682,64 +721,66 @@
- int res;
- switch(cond) {
- case AST_CONTROL_BUSY:
-- res = 1;
-- break;
- case AST_CONTROL_CONGESTION:
-- res = 2;
-- break;
- case AST_CONTROL_RINGING:
-- res = 0;
-+ res = cond;
- break;
- case -1:
-- cursound = -1;
-+ oss.cursound = -1;
- return 0;
- default:
- ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
- return -1;
- }
- if (res > -1) {
-- write(sndcmd[1], &res, sizeof(res));
-+ RING(res);
- }
- return 0;
- }
-
--static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
-+static struct ast_channel *oss_new(struct chan_oss_pvt *oss, int state)
- {
- struct ast_channel *tmp;
-+ struct ast_channel_pvt *pvt;
-+
- tmp = ast_channel_alloc(1);
-- if (tmp) {
-- snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
-- tmp->type = type;
-- tmp->fds[0] = sounddev;
-- tmp->nativeformats = AST_FORMAT_SLINEAR;
-- tmp->pvt->pvt = p;
-- tmp->pvt->send_digit = oss_digit;
-- tmp->pvt->send_text = oss_text;
-- tmp->pvt->hangup = oss_hangup;
-- tmp->pvt->answer = oss_answer;
-- tmp->pvt->read = oss_read;
-- tmp->pvt->call = oss_call;
-- tmp->pvt->write = oss_write;
-- tmp->pvt->indicate = oss_indicate;
-- tmp->pvt->fixup = oss_fixup;
-- if (strlen(p->context))
-- strncpy(tmp->context, p->context, sizeof(tmp->context)-1);
-- if (strlen(p->exten))
-- strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1);
-- if (strlen(language))
-- strncpy(tmp->language, language, sizeof(tmp->language)-1);
-- p->owner = tmp;
-- ast_setstate(tmp, state);
-- ast_mutex_lock(&usecnt_lock);
-- usecnt++;
-- ast_mutex_unlock(&usecnt_lock);
-- ast_update_use_count();
-- if (state != AST_STATE_DOWN) {
-- if (ast_pbx_start(tmp)) {
-- ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
-- ast_hangup(tmp);
-- tmp = NULL;
-- }
-+ if (tmp == NULL)
-+ return NULL;
-+ snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", oss->device + 5);
-+ tmp->type = type;
-+ tmp->fds[0] = oss->sounddev;
-+ tmp->nativeformats = AST_FORMAT_SLINEAR;
-+ pvt = tmp->pvt;
-+ pvt->pvt = oss;
-+#if 1
-+ pvt->send_digit = oss_digit;
-+ pvt->send_text = oss_text;
-+ pvt->hangup = oss_hangup;
-+ pvt->answer = oss_answer;
-+ pvt->read = oss_read;
-+ pvt->call = oss_call;
-+ pvt->write = oss_write;
-+ pvt->indicate = oss_indicate;
-+ pvt->fixup = oss_fixup;
-+#endif
-+ if (strlen(oss->context))
-+ strncpy(tmp->context, oss->context, sizeof(tmp->context)-1);
-+ if (strlen(oss->exten))
-+ strncpy(tmp->exten, oss->exten, sizeof(tmp->exten)-1);
-+ if (strlen(language))
-+ strncpy(tmp->language, language, sizeof(tmp->language)-1);
-+ oss->owner = tmp;
-+ ast_setstate(tmp, state);
-+ ast_mutex_lock(&usecnt_lock);
-+ usecnt++;
-+ ast_mutex_unlock(&usecnt_lock);
-+ ast_update_use_count();
-+ if (state != AST_STATE_DOWN) {
-+ if (ast_pbx_start(tmp)) {
-+ ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
-+ ast_hangup(tmp);
-+ tmp = NULL;
-+ /* XXX what about oss->owner and the channel itself ? */
- }
- }
- return tmp;
-@@ -770,13 +811,13 @@
- if ((argc != 1) && (argc != 2))
- return RESULT_SHOWUSAGE;
- if (argc == 1) {
-- ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
-+ ast_cli(fd, "Auto answer is %s.\n", oss.autoanswer ? "on" : "off");
- return RESULT_SUCCESS;
- } else {
- if (!strcasecmp(argv[1], "on"))
-- autoanswer = -1;
-+ oss.autoanswer = -1;
- else if (!strcasecmp(argv[1], "off"))
-- autoanswer = 0;
-+ oss.autoanswer = 0;
- else
- return RESULT_SHOWUSAGE;
- }
-@@ -788,12 +829,14 @@
- #ifndef MIN
- #define MIN(a,b) ((a) < (b) ? (a) : (b))
- #endif
-+ int l = strlen(word);
-+
- switch(state) {
- case 0:
-- if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
-+ if (l && !strncasecmp(word, "on", MIN(l, 2)))
- return strdup("on");
- case 1:
-- if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
-+ if (l && !strncasecmp(word, "off", MIN(l, 3)))
- return strdup("off");
- default:
- return NULL;
-@@ -816,8 +859,8 @@
- ast_cli(fd, "No one is calling us\n");
- return RESULT_FAILURE;
- }
-- hookstate = 1;
-- cursound = -1;
-+ oss.hookstate = 1;
-+ oss.cursound = -1;
- ast_queue_frame(oss.owner, &f);
- answer_sound();
- return RESULT_SUCCESS;
-@@ -863,12 +906,12 @@
- {
- if (argc != 1)
- return RESULT_SHOWUSAGE;
-- cursound = -1;
-- if (!oss.owner && !hookstate) {
-+ oss.cursound = -1;
-+ if (!oss.owner && !oss.hookstate) {
- ast_cli(fd, "No call to hangup up\n");
- return RESULT_FAILURE;
- }
-- hookstate = 0;
-+ oss.hookstate = 0;
- if (oss.owner) {
- ast_queue_hangup(oss.owner);
- }
-@@ -900,8 +943,8 @@
- }
- return RESULT_SUCCESS;
- }
-- mye = exten;
-- myc = context;
-+ mye = oss_exten;
-+ myc = default_context;
- if (argc == 2) {
- char *stringp=NULL;
- strncpy(tmp, argv[1], sizeof(tmp)-1);
-@@ -916,7 +959,7 @@
- if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
- strncpy(oss.exten, mye, sizeof(oss.exten)-1);
- strncpy(oss.context, myc, sizeof(oss.context)-1);
-- hookstate = 1;
-+ oss.hookstate = 1;
- oss_new(&oss, AST_STATE_RINGING);
- } else
- ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
-@@ -974,21 +1017,47 @@
- int res;
- int x;
- struct ast_config *cfg;
-- struct ast_variable *v;
-- res = pipe(sndcmd);
-+
-+ res = pipe(oss.sndcmd);
- if (res) {
- ast_log(LOG_ERROR, "Unable to create pipe\n");
- return -1;
- }
-- res = soundcard_init();
-- if (res < 0) {
-+ /* load config file */
-+ if ((cfg = ast_load(config))) {
-+ struct ast_variable *v = ast_variable_browse(cfg, "general");
-+ while(v) {
-+ if (!strcasecmp(v->name, "autoanswer"))
-+ oss.autoanswer = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "autohangup"))
-+ oss.autohangup = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "oss.silencesuppression"))
-+ oss.silencesuppression = ast_true(v->value);
-+ else if (!strcasecmp(v->name, "silencethreshold"))
-+ oss.silencethreshold = atoi(v->value);
-+ else if (!strcasecmp(v->name, "device"))
-+ strncpy(oss.device, v->value, sizeof(oss.device)-1);
-+ else if (!strcasecmp(v->name, "context"))
-+ strncpy(default_context, v->value, sizeof(default_context)-1);
-+ else if (!strcasecmp(v->name, "language"))
-+ strncpy(language, v->value, sizeof(language)-1);
-+ else if (!strcasecmp(v->name, "extension"))
-+ strncpy(oss_exten, v->value, sizeof(oss_exten)-1);
-+ v=v->next;
-+ }
-+ ast_destroy(cfg);
-+ }
-+ if (!strlen(oss.device))
-+ strncpy(oss.device, DEV_DSP, sizeof(oss.device)-1);
-+ if (setformat(&oss, O_RDWR) < 0) { /* open device */
- if (option_verbose > 1) {
- ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
- ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
- }
- return 0;
- }
-- if (!full_duplex)
-+ soundcard_setinput(1); /* force set if not full_duplex */
-+ if (oss.duplex != M_FULL)
- ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
- res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
- if (res < 0) {
-@@ -997,26 +1066,7 @@
- }
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_register(myclis + x);
-- if ((cfg = ast_load(config))) {
-- v = ast_variable_browse(cfg, "general");
-- while(v) {
-- if (!strcasecmp(v->name, "autoanswer"))
-- autoanswer = ast_true(v->value);
-- else if (!strcasecmp(v->name, "silencesuppression"))
-- silencesuppression = ast_true(v->value);
-- else if (!strcasecmp(v->name, "silencethreshold"))
-- silencethreshold = atoi(v->value);
-- else if (!strcasecmp(v->name, "context"))
-- strncpy(context, v->value, sizeof(context)-1);
-- else if (!strcasecmp(v->name, "language"))
-- strncpy(language, v->value, sizeof(language)-1);
-- else if (!strcasecmp(v->name, "extension"))
-- strncpy(exten, v->value, sizeof(exten)-1);
-- v=v->next;
-- }
-- ast_destroy(cfg);
-- }
-- ast_pthread_create(&sthread, NULL, sound_thread, NULL);
-+ ast_pthread_create(&oss.sthread, NULL, sound_thread, NULL);
- return 0;
- }
-
-@@ -1027,15 +1077,16 @@
- int x;
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_unregister(myclis + x);
-- close(sounddev);
-- if (sndcmd[0] > 0) {
-- close(sndcmd[0]);
-- close(sndcmd[1]);
-+ close(oss.sounddev);
-+ if (oss.sndcmd[0] > 0) {
-+ close(oss.sndcmd[0]);
-+ close(oss.sndcmd[1]);
- }
- if (oss.owner)
- ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
- if (oss.owner)
- return -1;
-+ /* XXX what about the thread ? */
- return 0;
- }
-
generated by cgit v1.2.3 (git 2.25.1) at 2025-09-07 21:06:39 +0000